2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
8 * Copyright 2015 Kurento (http://kurento.org/)
9 * @author: Miguel ParĂs <mparisdiaz@gmail.com>
11 * This library is free software; you can redistribute it and/or
12 * modify it under the terms of the GNU Library General Public
13 * License as published by the Free Software Foundation; either
14 * version 2 of the License, or (at your option) any later version.
16 * This library is distributed in the hope that it will be useful,
17 * but WITHOUT ANY WARRANTY; without even the implied warranty of
18 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19 * Library General Public License for more details.
21 * You should have received a copy of the GNU Library General Public
22 * License along with this library; if not, write to the
23 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
24 * Boston, MA 02110-1301, USA.
29 * SECTION:element-rtpjitterbuffer
31 * This element reorders and removes duplicate RTP packets as they are received
32 * from a network source.
34 * The element needs the clock-rate of the RTP payload in order to estimate the
35 * delay. This information is obtained either from the caps on the sink pad or,
36 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
37 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
39 * The rtpjitterbuffer will wait for missing packets up to a configurable time
40 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
41 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
42 * property is set, lost packets will result in a custom serialized downstream
43 * event of name GstRTPPacketLost. The lost packet events are usually used by a
44 * depayloader or other element to create concealment data or some other logic
45 * to gracefully handle the missing packets.
47 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
48 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
51 * The jitterbuffer can also be configured to send early retransmission events
52 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
53 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
54 * sends a custom upstream event named GstRTPRetransmissionRequest when the
55 * packet is considered late. The initial expected packet arrival time is
56 * calculated as follows:
58 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
59 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
60 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
61 * packets with different rtptime.
63 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
64 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
65 * previously scheduled timeout is overwritten.
67 * - If seqnum N arrived, all seqnum older than
68 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
69 * immediately. This is to request fast feedback for abonormally reorder
70 * packets before any of the previous timeouts is triggered.
72 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
73 * event. After the initial timeout expires and the retransmission event is
74 * sent, the timeout is scheduled for
75 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
76 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
77 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
78 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
79 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
80 * retransmission requests are sent and the regular logic is performed to
81 * schedule a lost packet as discussed above.
83 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
86 * This element will automatically be used inside rtpbin.
89 * <title>Example pipelines</title>
91 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
92 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
93 * inserted into the pipeline to smooth out network jitter and to reorder the
94 * out-of-order RTP packets.
105 #include <gst/rtp/gstrtpbuffer.h>
106 #include <gst/net/net.h>
108 #include "gstrtpjitterbuffer.h"
109 #include "rtpjitterbuffer.h"
110 #include "rtpstats.h"
112 #include <gst/glib-compat-private.h>
114 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
115 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
117 /* RTPJitterBuffer signals and args */
120 SIGNAL_REQUEST_PT_MAP,
128 #define DEFAULT_LATENCY_MS 200
129 #define DEFAULT_DROP_ON_LATENCY FALSE
130 #define DEFAULT_TS_OFFSET 0
131 #define DEFAULT_DO_LOST FALSE
132 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
133 #define DEFAULT_PERCENT 0
134 #define DEFAULT_DO_RETRANSMISSION FALSE
135 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
136 #define DEFAULT_RTX_DELAY -1
137 #define DEFAULT_RTX_MIN_DELAY 0
138 #define DEFAULT_RTX_DELAY_REORDER 3
139 #define DEFAULT_RTX_RETRY_TIMEOUT -1
140 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
141 #define DEFAULT_RTX_RETRY_PERIOD -1
142 #define DEFAULT_RTX_MAX_RETRIES -1
143 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
144 #define DEFAULT_MAX_DROPOUT_TIME 60000
145 #define DEFAULT_MAX_MISORDER_TIME 2000
146 #define DEFAULT_RFC7273_SYNC FALSE
148 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
149 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
155 PROP_DROP_ON_LATENCY,
160 PROP_DO_RETRANSMISSION,
161 PROP_RTX_NEXT_SEQNUM,
164 PROP_RTX_DELAY_REORDER,
165 PROP_RTX_RETRY_TIMEOUT,
166 PROP_RTX_MIN_RETRY_TIMEOUT,
167 PROP_RTX_RETRY_PERIOD,
168 PROP_RTX_MAX_RETRIES,
170 PROP_MAX_RTCP_RTP_TIME_DIFF,
171 PROP_MAX_DROPOUT_TIME,
172 PROP_MAX_MISORDER_TIME,
176 #define JBUF_LOCK(priv) G_STMT_START { \
177 GST_TRACE("Locking from thread %p", g_thread_self()); \
178 (g_mutex_lock (&(priv)->jbuf_lock)); \
179 GST_TRACE("Locked from thread %p", g_thread_self()); \
182 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
184 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
187 #define JBUF_UNLOCK(priv) G_STMT_START { \
188 GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
189 (g_mutex_unlock (&(priv)->jbuf_lock)); \
192 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
193 GST_DEBUG ("waiting timer"); \
194 (priv)->waiting_timer = TRUE; \
195 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
196 (priv)->waiting_timer = FALSE; \
197 GST_DEBUG ("waiting timer done"); \
199 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
200 if (G_UNLIKELY ((priv)->waiting_timer)) { \
201 GST_DEBUG ("signal timer"); \
202 g_cond_signal (&(priv)->jbuf_timer); \
206 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
207 GST_DEBUG ("waiting event"); \
208 (priv)->waiting_event = TRUE; \
209 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
210 (priv)->waiting_event = FALSE; \
211 GST_DEBUG ("waiting event done"); \
212 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
215 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
216 if (G_UNLIKELY ((priv)->waiting_event)) { \
217 GST_DEBUG ("signal event"); \
218 g_cond_signal (&(priv)->jbuf_event); \
222 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
223 GST_DEBUG ("waiting query"); \
224 (priv)->waiting_query = TRUE; \
225 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
226 (priv)->waiting_query = FALSE; \
227 GST_DEBUG ("waiting query done"); \
228 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
231 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
232 (priv)->last_query = res; \
233 if (G_UNLIKELY ((priv)->waiting_query)) { \
234 GST_DEBUG ("signal query"); \
235 g_cond_signal (&(priv)->jbuf_query); \
240 struct _GstRtpJitterBufferPrivate
242 GstPad *sinkpad, *srcpad;
245 RTPJitterBuffer *jbuf;
247 gboolean waiting_timer;
249 gboolean waiting_event;
251 gboolean waiting_query;
259 gboolean timer_running;
260 GThread *timer_thread;
265 gboolean drop_on_latency;
268 gboolean do_retransmission;
269 gboolean rtx_next_seqnum;
272 gint rtx_delay_reorder;
273 gint rtx_retry_timeout;
274 gint rtx_min_retry_timeout;
275 gint rtx_retry_period;
276 gint rtx_max_retries;
277 gint max_rtcp_rtp_time_diff;
278 guint32 max_dropout_time;
279 guint32 max_misorder_time;
281 /* the last seqnum we pushed out */
282 guint32 last_popped_seqnum;
283 /* the next expected seqnum we push */
285 /* seqnum-base, if known */
287 /* last output time */
288 GstClockTime last_out_time;
289 /* last valid input timestamp and rtptime pair */
290 GstClockTime ips_dts;
292 GstClockTime packet_spacing;
296 /* the next expected seqnum we receive */
297 GstClockTime last_in_dts;
298 guint32 next_in_seqnum;
302 /* start and stop ranges */
303 GstClockTime npt_start;
304 GstClockTime npt_stop;
305 guint64 ext_timestamp;
306 guint64 last_elapsed;
307 guint64 estimated_eos;
314 /* clock rate and rtp timestamp offset */
318 gint64 prev_ts_offset;
320 /* when we are shutting down */
321 GstFlowReturn srcresult;
327 GstClockTime timer_timeout;
328 guint16 timer_seqnum;
329 /* the latency of the upstream peer, we have to take this into account when
330 * synchronizing the buffers. */
331 GstClockTime peer_latency;
335 /* some accounting */
337 guint64 num_duplicates;
338 guint64 num_rtx_requests;
339 guint64 num_rtx_success;
340 guint64 num_rtx_failed;
343 RTPPacketRateCtx packet_rate_ctx;
346 GstClockTime last_dts;
347 guint64 last_rtptime;
348 GstClockTime avg_jitter;
365 GstClockTime timeout;
366 GstClockTime duration;
367 GstClockTime rtx_base;
368 GstClockTime rtx_delay;
369 GstClockTime rtx_retry;
370 GstClockTime rtx_last;
374 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
375 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
376 GstRtpJitterBufferPrivate))
378 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
379 GST_STATIC_PAD_TEMPLATE ("sink",
382 GST_STATIC_CAPS ("application/x-rtp"
383 /* "clock-rate = (int) [ 1, 2147483647 ], "
384 * "payload = (int) , "
385 * "encoding-name = (string) "
389 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
390 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
393 GST_STATIC_CAPS ("application/x-rtcp")
396 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
397 GST_STATIC_PAD_TEMPLATE ("src",
400 GST_STATIC_CAPS ("application/x-rtp"
401 /* "payload = (int) , "
402 * "clock-rate = (int) , "
403 * "encoding-name = (string) "
407 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
409 #define gst_rtp_jitter_buffer_parent_class parent_class
410 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
412 /* object overrides */
413 static void gst_rtp_jitter_buffer_set_property (GObject * object,
414 guint prop_id, const GValue * value, GParamSpec * pspec);
415 static void gst_rtp_jitter_buffer_get_property (GObject * object,
416 guint prop_id, GValue * value, GParamSpec * pspec);
417 static void gst_rtp_jitter_buffer_finalize (GObject * object);
419 /* element overrides */
420 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
421 * element, GstStateChange transition);
422 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
423 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
424 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
426 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
427 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
431 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
432 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
435 /* sinkpad overrides */
436 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
437 GstObject * parent, GstEvent * event);
438 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
439 GstObject * parent, GstBuffer * buffer);
441 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
442 GstObject * parent, GstEvent * event);
443 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
444 GstObject * parent, GstBuffer * buffer);
446 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
447 GstObject * parent, GstQuery * query);
449 /* srcpad overrides */
450 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
451 GstObject * parent, GstEvent * event);
452 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
453 GstObject * parent, GstPadMode mode, gboolean active);
454 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
455 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
456 GstObject * parent, GstQuery * query);
459 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
461 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
462 gboolean active, guint64 base_time);
463 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
465 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
466 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
468 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
470 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
474 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
476 GObjectClass *gobject_class;
477 GstElementClass *gstelement_class;
479 gobject_class = (GObjectClass *) klass;
480 gstelement_class = (GstElementClass *) klass;
482 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
484 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
486 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
487 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
490 * GstRtpJitterBuffer:latency:
492 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
493 * for at most this time.
495 g_object_class_install_property (gobject_class, PROP_LATENCY,
496 g_param_spec_uint ("latency", "Buffer latency in ms",
497 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
498 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
500 * GstRtpJitterBuffer:drop-on-latency:
502 * Drop oldest buffers when the queue is completely filled.
504 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
505 g_param_spec_boolean ("drop-on-latency",
506 "Drop buffers when maximum latency is reached",
507 "Tells the jitterbuffer to never exceed the given latency in size",
508 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
510 * GstRtpJitterBuffer:ts-offset:
512 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
513 * This is mainly used to ensure interstream synchronisation.
515 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
516 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
517 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
518 G_MAXINT64, DEFAULT_TS_OFFSET,
519 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
522 * GstRtpJitterBuffer:do-lost:
524 * Send out a GstRTPPacketLost event downstream when a packet is considered
527 g_object_class_install_property (gobject_class, PROP_DO_LOST,
528 g_param_spec_boolean ("do-lost", "Do Lost",
529 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
530 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
533 * GstRtpJitterBuffer:mode:
535 * Control the buffering and timestamping mode used by the jitterbuffer.
537 g_object_class_install_property (gobject_class, PROP_MODE,
538 g_param_spec_enum ("mode", "Mode",
539 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
540 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
542 * GstRtpJitterBuffer:percent:
544 * The percent of the jitterbuffer that is filled.
546 g_object_class_install_property (gobject_class, PROP_PERCENT,
547 g_param_spec_int ("percent", "percent",
548 "The buffer filled percent", 0, 100,
549 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
551 * GstRtpJitterBuffer:do-retransmission:
553 * Send out a GstRTPRetransmission event upstream when a packet is considered
554 * late and should be retransmitted.
558 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
559 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
560 "Send retransmission events upstream when a packet is late",
561 DEFAULT_DO_RETRANSMISSION,
562 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
565 * GstRtpJitterBuffer:rtx-next-seqnum
567 * Estimate when the next packet should arrive and schedule a retransmission
569 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
570 * for packet N+1. So it will be requested if it does not arrive at the expected time.
571 * The expected time is calculated using the dts of N and the packet spacing.
575 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
576 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
577 "Estimate when the next packet should arrive and schedule a "
578 "retransmission request for it.",
579 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
582 * GstRtpJitterBuffer:rtx-delay:
584 * When a packet did not arrive at the expected time, wait this extra amount
585 * of time before sending a retransmission event.
587 * When -1 is used, the max jitter will be used as extra delay.
591 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
592 g_param_spec_int ("rtx-delay", "RTX Delay",
593 "Extra time in ms to wait before sending retransmission "
594 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
595 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
598 * GstRtpJitterBuffer:rtx-min-delay:
600 * When a packet did not arrive at the expected time, wait at least this extra amount
601 * of time before sending a retransmission event.
605 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
606 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
607 "Minimum time in ms to wait before sending retransmission "
608 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
609 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
611 * GstRtpJitterBuffer:rtx-delay-reorder:
613 * Assume that a retransmission event should be sent when we see
614 * this much packet reordering.
616 * When -1 is used, the value will be estimated based on observed packet
617 * reordering. When 0 is used packet reordering alone will not cause a
618 * retransmission event (Since 1.10).
622 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
623 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
624 "Sending retransmission event when this much reordering "
625 "(0 disable, -1 automatic)",
626 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
627 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
629 * GstRtpJitterBuffer::rtx-retry-timeout:
631 * When no packet has been received after sending a retransmission event
632 * for this time, retry sending a retransmission event.
634 * When -1 is used, the value will be estimated based on observed round
639 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
640 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
641 "Retry sending a transmission event after this timeout in "
642 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
643 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
645 * GstRtpJitterBuffer::rtx-min-retry-timeout:
647 * The minimum amount of time between retry timeouts. When
648 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
649 * minimum interval between retry timeouts.
651 * When -1 is used, the value will be estimated based on the
656 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
657 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
658 "Minimum timeout between sending a transmission event in "
659 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
660 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
662 * GstRtpJitterBuffer:rtx-retry-period:
664 * The amount of time to try to get a retransmission.
666 * When -1 is used, the value will be estimated based on the jitterbuffer
667 * latency and the observed round trip time.
671 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
672 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
673 "Try to get a retransmission for this many ms "
674 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
675 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
677 * GstRtpJitterBuffer:rtx-max-retries:
679 * The maximum number of retries to request a retransmission.
681 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
682 * When -1 is used, the number of retransmission request will not be limited.
686 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
687 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
688 "The maximum number of retries to request a retransmission. "
689 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
690 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
692 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
693 g_param_spec_uint ("max-dropout-time", "Max dropout time",
694 "The maximum time (milliseconds) of missing packets tolerated.",
695 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
696 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
698 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
699 g_param_spec_uint ("max-misorder-time", "Max misorder time",
700 "The maximum time (milliseconds) of misordered packets tolerated.",
701 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
702 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
704 * GstRtpJitterBuffer:stats:
706 * Various jitterbuffer statistics. This property returns a GstStructure
707 * with name application/x-rtp-jitterbuffer-stats with the following fields:
713 * <classname>"rtx-count"</classname>:
714 * the number of retransmissions requested.
720 * <classname>"rtx-success-count"</classname>:
721 * the number of successful retransmissions.
727 * <classname>"rtx-per-packet"</classname>:
728 * average number of RTX per packet.
734 * <classname>"rtx-rtt"</classname>:
735 * average round trip time per RTX.
742 g_object_class_install_property (gobject_class, PROP_STATS,
743 g_param_spec_boxed ("stats", "Statistics",
744 "Various statistics", GST_TYPE_STRUCTURE,
745 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
748 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
750 * The maximum amount of time in ms that the RTP time in the RTCP SRs
751 * is allowed to be ahead of the last RTP packet we received. Use
752 * -1 to disable ignoring of RTCP packets.
756 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
757 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
758 "Maximum amount of time in ms that the RTP time in RTCP SRs "
759 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
760 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
761 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
763 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
764 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
765 "Synchronize received streams to the RFC7273 clock "
766 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
767 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
770 * GstRtpJitterBuffer::request-pt-map:
771 * @buffer: the object which received the signal
774 * Request the payload type as #GstCaps for @pt.
776 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
777 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
778 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
779 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
780 GST_TYPE_CAPS, 1, G_TYPE_UINT);
782 * GstRtpJitterBuffer::handle-sync:
783 * @buffer: the object which received the signal
784 * @struct: a GstStructure containing sync values.
786 * Be notified of new sync values.
788 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
789 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
790 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
791 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
792 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
795 * GstRtpJitterBuffer::on-npt-stop:
796 * @buffer: the object which received the signal
798 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
799 * the npt-stop position.
801 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
802 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
803 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
804 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
805 G_TYPE_NONE, 0, G_TYPE_NONE);
808 * GstRtpJitterBuffer::clear-pt-map:
809 * @buffer: the object which received the signal
811 * Invalidate the clock-rate as obtained with the
812 * #GstRtpJitterBuffer::request-pt-map signal.
814 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
815 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
816 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
817 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
818 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
821 * GstRtpJitterBuffer::set-active:
822 * @buffer: the object which received the signal
824 * Start pushing out packets with the given base time. This signal is only
825 * useful in buffering mode.
827 * Returns: the time of the last pushed packet.
829 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
830 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
831 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
832 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
833 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
836 gstelement_class->change_state =
837 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
838 gstelement_class->request_new_pad =
839 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
840 gstelement_class->release_pad =
841 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
842 gstelement_class->provide_clock =
843 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
844 gstelement_class->set_clock =
845 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
847 gst_element_class_add_static_pad_template (gstelement_class,
848 &gst_rtp_jitter_buffer_src_template);
849 gst_element_class_add_static_pad_template (gstelement_class,
850 &gst_rtp_jitter_buffer_sink_template);
851 gst_element_class_add_static_pad_template (gstelement_class,
852 &gst_rtp_jitter_buffer_sink_rtcp_template);
854 gst_element_class_set_static_metadata (gstelement_class,
855 "RTP packet jitter-buffer", "Filter/Network/RTP",
856 "A buffer that deals with network jitter and other transmission faults",
857 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
858 "Wim Taymans <wim.taymans@gmail.com>");
860 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
861 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
863 GST_DEBUG_CATEGORY_INIT
864 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
868 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
870 GstRtpJitterBufferPrivate *priv;
872 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
873 jitterbuffer->priv = priv;
875 priv->latency_ms = DEFAULT_LATENCY_MS;
876 priv->latency_ns = priv->latency_ms * GST_MSECOND;
877 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
878 priv->do_lost = DEFAULT_DO_LOST;
879 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
880 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
881 priv->rtx_delay = DEFAULT_RTX_DELAY;
882 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
883 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
884 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
885 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
886 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
887 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
888 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
889 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
890 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
893 priv->last_rtptime = -1;
894 priv->avg_jitter = 0;
895 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
896 priv->jbuf = rtp_jitter_buffer_new ();
897 g_mutex_init (&priv->jbuf_lock);
898 g_cond_init (&priv->jbuf_timer);
899 g_cond_init (&priv->jbuf_event);
900 g_cond_init (&priv->jbuf_query);
901 g_queue_init (&priv->gap_packets);
902 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
904 /* reset skew detection initialy */
905 rtp_jitter_buffer_reset_skew (priv->jbuf);
906 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
907 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
911 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
914 gst_pad_set_activatemode_function (priv->srcpad,
915 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
916 gst_pad_set_query_function (priv->srcpad,
917 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
918 gst_pad_set_event_function (priv->srcpad,
919 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
922 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
925 gst_pad_set_chain_function (priv->sinkpad,
926 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
927 gst_pad_set_event_function (priv->sinkpad,
928 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
929 gst_pad_set_query_function (priv->sinkpad,
930 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
932 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
933 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
935 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
938 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
940 #define ITEM_TYPE_BUFFER 0
941 #define ITEM_TYPE_LOST 1
942 #define ITEM_TYPE_EVENT 2
943 #define ITEM_TYPE_QUERY 3
945 static RTPJitterBufferItem *
946 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
947 guint seqnum, guint count, guint rtptime)
949 RTPJitterBufferItem *item;
951 item = g_slice_new (RTPJitterBufferItem);
958 item->seqnum = seqnum;
960 item->rtptime = rtptime;
966 free_item (RTPJitterBufferItem * item)
968 g_return_if_fail (item != NULL);
970 if (item->data && item->type != ITEM_TYPE_QUERY)
971 gst_mini_object_unref (item->data);
972 g_slice_free (RTPJitterBufferItem, item);
976 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
978 GList **l = user_data;
980 if (item->data && item->type == ITEM_TYPE_EVENT
981 && GST_EVENT_IS_STICKY (item->data)) {
982 *l = g_list_prepend (*l, item->data);
983 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
984 gst_mini_object_unref (item->data);
986 g_slice_free (RTPJitterBufferItem, item);
990 gst_rtp_jitter_buffer_finalize (GObject * object)
992 GstRtpJitterBuffer *jitterbuffer;
993 GstRtpJitterBufferPrivate *priv;
995 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
996 priv = jitterbuffer->priv;
998 g_array_free (priv->timers, TRUE);
999 g_mutex_clear (&priv->jbuf_lock);
1000 g_cond_clear (&priv->jbuf_timer);
1001 g_cond_clear (&priv->jbuf_event);
1002 g_cond_clear (&priv->jbuf_query);
1004 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1005 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1006 g_queue_clear (&priv->gap_packets);
1007 g_object_unref (priv->jbuf);
1009 G_OBJECT_CLASS (parent_class)->finalize (object);
1012 static GstIterator *
1013 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1015 GstRtpJitterBuffer *jitterbuffer;
1016 GstPad *otherpad = NULL;
1017 GstIterator *it = NULL;
1018 GValue val = { 0, };
1020 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1022 if (pad == jitterbuffer->priv->sinkpad) {
1023 otherpad = jitterbuffer->priv->srcpad;
1024 } else if (pad == jitterbuffer->priv->srcpad) {
1025 otherpad = jitterbuffer->priv->sinkpad;
1026 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1027 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1031 g_value_init (&val, GST_TYPE_PAD);
1032 g_value_set_object (&val, otherpad);
1033 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1034 g_value_unset (&val);
1041 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1043 GstRtpJitterBufferPrivate *priv;
1045 priv = jitterbuffer->priv;
1047 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1050 gst_pad_new_from_static_template
1051 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1052 gst_pad_set_chain_function (priv->rtcpsinkpad,
1053 gst_rtp_jitter_buffer_chain_rtcp);
1054 gst_pad_set_event_function (priv->rtcpsinkpad,
1055 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1056 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1057 gst_rtp_jitter_buffer_iterate_internal_links);
1058 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1059 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1061 return priv->rtcpsinkpad;
1065 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1067 GstRtpJitterBufferPrivate *priv;
1069 priv = jitterbuffer->priv;
1071 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1073 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1075 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1076 priv->rtcpsinkpad = NULL;
1080 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1081 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1083 GstRtpJitterBuffer *jitterbuffer;
1084 GstElementClass *klass;
1086 GstRtpJitterBufferPrivate *priv;
1088 g_return_val_if_fail (templ != NULL, NULL);
1089 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1091 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1092 priv = jitterbuffer->priv;
1093 klass = GST_ELEMENT_GET_CLASS (element);
1095 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1097 /* figure out the template */
1098 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1099 if (priv->rtcpsinkpad != NULL)
1102 result = create_rtcp_sink (jitterbuffer);
1104 goto wrong_template;
1111 g_warning ("rtpjitterbuffer: this is not our template");
1116 g_warning ("rtpjitterbuffer: pad already requested");
1122 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1124 GstRtpJitterBuffer *jitterbuffer;
1125 GstRtpJitterBufferPrivate *priv;
1127 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1128 g_return_if_fail (GST_IS_PAD (pad));
1130 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1131 priv = jitterbuffer->priv;
1133 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1135 if (priv->rtcpsinkpad == pad) {
1136 remove_rtcp_sink (jitterbuffer);
1145 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1151 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1153 return gst_system_clock_obtain ();
1157 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1159 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1161 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1163 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1167 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1169 GstRtpJitterBufferPrivate *priv;
1171 priv = jitterbuffer->priv;
1173 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1176 priv->clock_rate = -1;
1177 /* do not clear current content, but refresh state for new arrival */
1178 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1179 rtp_jitter_buffer_reset_skew (priv->jbuf);
1184 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1187 GstRtpJitterBufferPrivate *priv;
1188 GstClockTime last_out;
1189 RTPJitterBufferItem *item;
1194 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1195 active, GST_TIME_ARGS (offset));
1197 if (active != priv->active) {
1198 /* add the amount of time spent in paused to the output offset. All
1199 * outgoing buffers will have this offset applied to their timestamps in
1200 * order to make them arrive in time in the sink. */
1201 priv->out_offset = offset;
1202 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1203 GST_TIME_ARGS (priv->out_offset));
1204 priv->active = active;
1205 JBUF_SIGNAL_EVENT (priv);
1208 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1210 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1211 /* head buffer timestamp and offset gives our output time */
1212 last_out = item->dts + priv->ts_offset;
1214 /* use last known time when the buffer is empty */
1215 last_out = priv->last_out_time;
1223 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1225 GstRtpJitterBuffer *jitterbuffer;
1226 GstRtpJitterBufferPrivate *priv;
1231 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1232 priv = jitterbuffer->priv;
1234 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1236 caps = gst_pad_peer_query_caps (other, filter);
1238 templ = gst_pad_get_pad_template_caps (pad);
1240 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1245 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1247 intersect = gst_caps_intersect (caps, templ);
1248 gst_caps_unref (caps);
1249 gst_caps_unref (templ);
1253 gst_object_unref (jitterbuffer);
1259 * Must be called with JBUF_LOCK held
1263 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1264 GstCaps * caps, gint pt)
1266 GstRtpJitterBufferPrivate *priv;
1267 GstStructure *caps_struct;
1271 const gchar *ts_refclk, *mediaclk;
1273 priv = jitterbuffer->priv;
1275 /* first parse the caps */
1276 caps_struct = gst_caps_get_structure (caps, 0);
1278 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1280 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1282 GST_ERROR_OBJECT (jitterbuffer,
1283 "Got caps with wrong payload type (got %d, expected %d)", payload, pt);
1287 if (payload != -1) {
1288 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1289 priv->last_pt = payload;
1292 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1293 * measure the amount of data in the buffer */
1294 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1297 if (priv->clock_rate <= 0)
1300 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1302 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1304 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1306 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1307 * can use this to track the amount of time elapsed on the sender. */
1308 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1309 priv->clock_base = val;
1311 priv->clock_base = -1;
1313 priv->ext_timestamp = priv->clock_base;
1315 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1318 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1319 /* first expected seqnum, only update when we didn't have a previous base. */
1320 if (priv->next_in_seqnum == -1)
1321 priv->next_in_seqnum = val;
1322 if (priv->next_seqnum == -1) {
1323 priv->next_seqnum = val;
1324 JBUF_SIGNAL_EVENT (priv);
1326 priv->seqnum_base = val;
1328 priv->seqnum_base = -1;
1331 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1333 /* the start and stop times. The seqnum-base corresponds to the start time. We
1334 * will keep track of the seqnums on the output and when we reach the one
1335 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1336 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1337 priv->npt_start = tval;
1339 priv->npt_start = 0;
1341 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1342 priv->npt_stop = tval;
1344 priv->npt_stop = -1;
1346 GST_DEBUG_OBJECT (jitterbuffer,
1347 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1348 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1350 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1351 GstClock *clock = NULL;
1352 guint64 clock_offset = -1;
1354 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1357 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1358 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1359 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1361 const gchar *host, *portstr;
1365 host = ts_refclk + sizeof ("ntp=") - 1;
1366 if (host[0] == '[') {
1368 portstr = strchr (host, ']');
1369 if (portstr && portstr[1] == ':')
1370 portstr = portstr + 1;
1374 portstr = strrchr (host, ':');
1378 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1382 hostname = g_strndup (host, (portstr - host));
1384 hostname = g_strdup (host);
1386 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1389 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1390 const gchar *domainstr =
1391 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1394 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1397 clock = gst_ptp_clock_new (NULL, domain);
1399 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1402 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1403 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1405 if (!g_str_has_prefix (mediaclk, "direct=")
1406 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1407 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1408 if (strstr (mediaclk, "rate=") != NULL) {
1409 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1414 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1416 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1424 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1429 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1435 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1437 GstRtpJitterBufferPrivate *priv;
1439 priv = jitterbuffer->priv;
1442 /* mark ourselves as flushing */
1443 priv->srcresult = GST_FLOW_FLUSHING;
1444 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1445 /* this unblocks any waiting pops on the src pad task */
1446 JBUF_SIGNAL_EVENT (priv);
1447 JBUF_SIGNAL_QUERY (priv, FALSE);
1452 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1454 GstRtpJitterBufferPrivate *priv;
1456 priv = jitterbuffer->priv;
1459 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1460 /* Mark as non flushing */
1461 priv->srcresult = GST_FLOW_OK;
1462 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1463 priv->last_popped_seqnum = -1;
1464 priv->last_out_time = -1;
1465 priv->next_seqnum = -1;
1466 priv->seqnum_base = -1;
1467 priv->ips_rtptime = -1;
1468 priv->ips_dts = GST_CLOCK_TIME_NONE;
1469 priv->packet_spacing = 0;
1470 priv->next_in_seqnum = -1;
1471 priv->clock_rate = -1;
1474 priv->estimated_eos = -1;
1475 priv->last_elapsed = 0;
1476 priv->ext_timestamp = -1;
1477 priv->avg_jitter = 0;
1478 priv->last_dts = -1;
1479 priv->last_rtptime = -1;
1480 priv->last_in_dts = 0;
1481 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1482 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1483 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1484 rtp_jitter_buffer_reset_skew (priv->jbuf);
1485 remove_all_timers (jitterbuffer);
1486 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1487 g_queue_clear (&priv->gap_packets);
1492 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1493 GstPadMode mode, gboolean active)
1496 GstRtpJitterBuffer *jitterbuffer = NULL;
1498 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1501 case GST_PAD_MODE_PUSH:
1503 /* allow data processing */
1504 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1506 /* start pushing out buffers */
1507 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1508 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1509 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1511 /* make sure all data processing stops ASAP */
1512 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1514 /* NOTE this will hardlock if the state change is called from the src pad
1515 * task thread because we will _join() the thread. */
1516 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1517 result = gst_pad_stop_task (pad);
1527 static GstStateChangeReturn
1528 gst_rtp_jitter_buffer_change_state (GstElement * element,
1529 GstStateChange transition)
1531 GstRtpJitterBuffer *jitterbuffer;
1532 GstRtpJitterBufferPrivate *priv;
1533 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1535 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1536 priv = jitterbuffer->priv;
1538 switch (transition) {
1539 case GST_STATE_CHANGE_NULL_TO_READY:
1541 case GST_STATE_CHANGE_READY_TO_PAUSED:
1543 /* reset negotiated values */
1544 priv->clock_rate = -1;
1545 priv->clock_base = -1;
1546 priv->peer_latency = 0;
1548 /* block until we go to PLAYING */
1549 priv->blocked = TRUE;
1550 priv->timer_running = TRUE;
1551 priv->timer_thread =
1552 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1555 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1557 /* unblock to allow streaming in PLAYING */
1558 priv->blocked = FALSE;
1559 JBUF_SIGNAL_EVENT (priv);
1560 JBUF_SIGNAL_TIMER (priv);
1567 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1569 switch (transition) {
1570 case GST_STATE_CHANGE_READY_TO_PAUSED:
1571 /* we are a live element because we sync to the clock, which we can only
1572 * do in the PLAYING state */
1573 if (ret != GST_STATE_CHANGE_FAILURE)
1574 ret = GST_STATE_CHANGE_NO_PREROLL;
1576 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1578 /* block to stop streaming when PAUSED */
1579 priv->blocked = TRUE;
1580 unschedule_current_timer (jitterbuffer);
1582 if (ret != GST_STATE_CHANGE_FAILURE)
1583 ret = GST_STATE_CHANGE_NO_PREROLL;
1585 case GST_STATE_CHANGE_PAUSED_TO_READY:
1587 gst_buffer_replace (&priv->last_sr, NULL);
1588 priv->timer_running = FALSE;
1589 unschedule_current_timer (jitterbuffer);
1590 JBUF_SIGNAL_TIMER (priv);
1591 JBUF_SIGNAL_QUERY (priv, FALSE);
1593 g_thread_join (priv->timer_thread);
1594 priv->timer_thread = NULL;
1596 case GST_STATE_CHANGE_READY_TO_NULL:
1606 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1609 gboolean ret = TRUE;
1610 GstRtpJitterBuffer *jitterbuffer;
1611 GstRtpJitterBufferPrivate *priv;
1613 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1614 priv = jitterbuffer->priv;
1616 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1618 switch (GST_EVENT_TYPE (event)) {
1619 case GST_EVENT_LATENCY:
1621 GstClockTime latency;
1623 gst_event_parse_latency (event, &latency);
1625 GST_DEBUG_OBJECT (jitterbuffer,
1626 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1629 /* adjust the overall buffer delay to the total pipeline latency in
1630 * buffering mode because if downstream consumes too fast (because of
1631 * large latency or queues, we would start rebuffering again. */
1632 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1633 RTP_JITTER_BUFFER_MODE_BUFFER) {
1634 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1638 ret = gst_pad_push_event (priv->sinkpad, event);
1642 ret = gst_pad_push_event (priv->sinkpad, event);
1649 /* handles and stores the event in the jitterbuffer, must be called with
1652 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1654 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1655 RTPJitterBufferItem *item;
1658 switch (GST_EVENT_TYPE (event)) {
1659 case GST_EVENT_CAPS:
1663 gst_event_parse_caps (event, &caps);
1664 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1667 case GST_EVENT_SEGMENT:
1670 gst_event_copy_segment (event, &segment);
1672 /* we need time for now */
1673 if (segment.format != GST_FORMAT_TIME) {
1674 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1675 gst_event_unref (event);
1677 gst_segment_init (&segment, GST_FORMAT_TIME);
1678 event = gst_event_new_segment (&segment);
1681 priv->segment = segment;
1686 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1693 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1694 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1695 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
1697 JBUF_SIGNAL_EVENT (priv);
1703 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1706 gboolean ret = TRUE;
1707 GstRtpJitterBuffer *jitterbuffer;
1708 GstRtpJitterBufferPrivate *priv;
1710 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1711 priv = jitterbuffer->priv;
1713 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1715 switch (GST_EVENT_TYPE (event)) {
1716 case GST_EVENT_FLUSH_START:
1717 ret = gst_pad_push_event (priv->srcpad, event);
1718 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1719 /* wait for the loop to go into PAUSED */
1720 gst_pad_pause_task (priv->srcpad);
1722 case GST_EVENT_FLUSH_STOP:
1723 ret = gst_pad_push_event (priv->srcpad, event);
1725 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1726 GST_PAD_MODE_PUSH, TRUE);
1729 if (GST_EVENT_IS_SERIALIZED (event)) {
1730 /* serialized events go in the queue */
1732 if (priv->srcresult != GST_FLOW_OK) {
1733 /* Errors in sticky event pushing are no problem and ignored here
1734 * as they will cause more meaningful errors during data flow.
1735 * For EOS events, that are not followed by data flow, we still
1736 * return FALSE here though.
1738 if (!GST_EVENT_IS_STICKY (event) ||
1739 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1740 goto out_flow_error;
1742 /* refuse more events on EOS */
1745 ret = queue_event (jitterbuffer, event);
1748 /* non-serialized events are forwarded downstream immediately */
1749 ret = gst_pad_push_event (priv->srcpad, event);
1758 GST_DEBUG_OBJECT (jitterbuffer,
1759 "refusing event, we have a downstream flow error: %s",
1760 gst_flow_get_name (priv->srcresult));
1762 gst_event_unref (event);
1767 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1769 gst_event_unref (event);
1775 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1778 gboolean ret = TRUE;
1779 GstRtpJitterBuffer *jitterbuffer;
1781 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1783 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1785 switch (GST_EVENT_TYPE (event)) {
1786 case GST_EVENT_FLUSH_START:
1787 gst_event_unref (event);
1789 case GST_EVENT_FLUSH_STOP:
1790 gst_event_unref (event);
1793 ret = gst_pad_event_default (pad, parent, event);
1801 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1802 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1803 * GST_FLOW_FLUSHING when the element is shutting down. On success
1804 * GST_FLOW_OK is returned.
1806 static GstFlowReturn
1807 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1811 GValue args[2] = { {0}, {0} };
1815 g_value_init (&args[0], GST_TYPE_ELEMENT);
1816 g_value_set_object (&args[0], jitterbuffer);
1817 g_value_init (&args[1], G_TYPE_UINT);
1818 g_value_set_uint (&args[1], pt);
1820 g_value_init (&ret, GST_TYPE_CAPS);
1821 g_value_set_boxed (&ret, NULL);
1823 JBUF_UNLOCK (jitterbuffer->priv);
1824 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1826 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1828 g_value_unset (&args[0]);
1829 g_value_unset (&args[1]);
1830 caps = (GstCaps *) g_value_dup_boxed (&ret);
1831 g_value_unset (&ret);
1835 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1836 gst_caps_unref (caps);
1838 if (G_UNLIKELY (!res))
1846 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1847 return GST_FLOW_ERROR;
1851 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1852 return GST_FLOW_FLUSHING;
1856 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1857 return GST_FLOW_ERROR;
1861 /* call with jbuf lock held */
1863 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1865 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1866 GstMessage *message = NULL;
1871 /* Post a buffering message */
1872 if (priv->last_percent != percent) {
1873 priv->last_percent = percent;
1875 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1876 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1883 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1885 GstRtpJitterBufferPrivate *priv;
1887 priv = jitterbuffer->priv;
1889 if (timestamp == -1)
1892 /* apply the timestamp offset, this is used for inter stream sync */
1893 timestamp += priv->ts_offset;
1894 /* add the offset, this is used when buffering */
1895 timestamp += priv->out_offset;
1901 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
1903 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1904 TimerData *timer = NULL;
1907 len = priv->timers->len;
1908 for (i = 0; i < len; i++) {
1909 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1910 if (test->seqnum == seqnum) {
1919 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1921 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1923 if (priv->clock_id) {
1924 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1925 gst_clock_id_unschedule (priv->clock_id);
1926 priv->clock_id = NULL;
1931 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1933 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1934 GstClockTime test_timeout;
1936 if ((test_timeout = timer->timeout) == -1)
1939 if (timer->type != TIMER_TYPE_EXPECTED) {
1940 /* add our latency and offset to get output times. */
1941 test_timeout = apply_offset (jitterbuffer, test_timeout);
1942 test_timeout += priv->latency_ns;
1944 return test_timeout;
1948 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1950 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1952 if (priv->clock_id) {
1953 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1955 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1956 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1958 if (timeout == -1 || timeout < priv->timer_timeout)
1959 unschedule_current_timer (jitterbuffer);
1964 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1965 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1966 GstClockTime duration)
1968 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1972 GST_DEBUG_OBJECT (jitterbuffer,
1973 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1974 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1975 GST_TIME_ARGS (delay));
1977 len = priv->timers->len;
1978 g_array_set_size (priv->timers, len + 1);
1979 timer = &g_array_index (priv->timers, TimerData, len);
1982 timer->seqnum = seqnum;
1984 timer->timeout = timeout + delay;
1985 timer->duration = duration;
1986 if (type == TIMER_TYPE_EXPECTED) {
1987 timer->rtx_base = timeout;
1988 timer->rtx_delay = delay;
1989 timer->rtx_retry = 0;
1991 timer->num_rtx_retry = 0;
1992 recalculate_timer (jitterbuffer, timer);
1993 JBUF_SIGNAL_TIMER (priv);
1999 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2000 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2002 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2003 gboolean seqchange, timechange;
2006 seqchange = timer->seqnum != seqnum;
2007 timechange = timer->timeout != timeout;
2009 if (!seqchange && !timechange)
2012 oldseq = timer->seqnum;
2014 GST_DEBUG_OBJECT (jitterbuffer,
2015 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
2016 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
2018 timer->timeout = timeout + delay;
2019 timer->seqnum = seqnum;
2021 timer->rtx_base = timeout;
2022 timer->rtx_delay = delay;
2023 timer->rtx_retry = 0;
2026 timer->num_rtx_retry = 0;
2028 if (priv->clock_id) {
2029 /* we changed the seqnum and there is a timer currently waiting with this
2030 * seqnum, unschedule it */
2031 if (seqchange && priv->timer_seqnum == oldseq)
2032 unschedule_current_timer (jitterbuffer);
2033 /* we changed the time, check if it is earlier than what we are waiting
2034 * for and unschedule if so */
2035 else if (timechange)
2036 recalculate_timer (jitterbuffer, timer);
2041 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2042 guint16 seqnum, GstClockTime timeout)
2046 /* find the seqnum timer */
2047 timer = find_timer (jitterbuffer, seqnum);
2048 if (timer == NULL) {
2049 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2051 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2057 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2059 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2062 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2063 unschedule_current_timer (jitterbuffer);
2066 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2067 g_array_remove_index_fast (priv->timers, idx);
2072 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2074 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2075 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2076 g_array_set_size (priv->timers, 0);
2077 unschedule_current_timer (jitterbuffer);
2080 /* get the extra delay to wait before sending RTX */
2082 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2086 if (priv->rtx_delay == -1) {
2087 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2088 delay = DEFAULT_AUTO_RTX_DELAY;
2090 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2091 * packet spacing is a good margin */
2092 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2095 delay = priv->rtx_delay * GST_MSECOND;
2097 if (priv->rtx_min_delay > 0)
2098 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2103 /* Check if packet with seqnum is already considered definitely lost by being
2104 * part of a "lost timer" for multiple packets */
2106 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2108 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2111 len = priv->timers->len;
2112 for (i = 0; i < len; i++) {
2113 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2114 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2116 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2118 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2119 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2127 /* we just received a packet with seqnum and dts.
2129 * First check for old seqnum that we are still expecting. If the gap with the
2130 * current seqnum is too big, unschedule the timeouts.
2132 * If we have a valid packet spacing estimate we can set a timer for when we
2133 * should receive the next packet.
2134 * If we don't have a valid estimate, we remove any timer we might have
2135 * had for this packet.
2138 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2139 GstClockTime dts, gboolean do_next_seqnum)
2141 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2142 TimerData *timer = NULL;
2144 /* go through all timers and unschedule the ones with a large gap */
2145 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2147 len = priv->timers->len;
2148 for (i = 0; i < len; i++) {
2149 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2152 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2154 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2155 test->type, test->seqnum, seqnum, gap);
2158 GST_DEBUG ("found timer for current seqnum");
2159 /* the timer for the current seqnum */
2161 } else if (gap > priv->rtx_delay_reorder) {
2162 /* max gap, we exceeded the max reorder distance and we don't expect the
2163 * missing packet to be this reordered */
2164 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2165 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2169 /* find the timer for the seqnum */
2170 timer = find_timer (jitterbuffer, seqnum);
2173 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2174 && priv->do_retransmission && priv->rtx_next_seqnum;
2176 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2177 if (timer->num_rtx_retry > 0) {
2178 GstClockTime rtx_last, delay;
2180 /* we scheduled a retry for this packet and now we have it */
2181 priv->num_rtx_success++;
2182 /* all the previous retry attempts failed */
2183 priv->num_rtx_failed += timer->num_rtx_retry - 1;
2184 /* number of retries before receiving the packet */
2185 if (priv->avg_rtx_num == 0.0)
2186 priv->avg_rtx_num = timer->num_rtx_retry;
2188 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2189 /* calculate the delay between retransmission request and receiving this
2190 * packet, start with when we scheduled this timeout last */
2191 rtx_last = timer->rtx_last;
2192 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2193 /* we have a valid delay if this packet arrived after we scheduled the
2195 delay = dts - rtx_last;
2196 if (priv->avg_rtx_rtt == 0)
2197 priv->avg_rtx_rtt = delay;
2199 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2203 GST_LOG_OBJECT (jitterbuffer,
2204 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2205 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2206 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2207 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2208 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2209 GST_TIME_ARGS (priv->avg_rtx_rtt));
2211 /* don't try to estimate the next seqnum because this is a retransmitted
2212 * packet and it probably did not arrive with the expected packet
2214 do_next_seqnum = FALSE;
2218 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2219 GstClockTime expected, delay;
2221 /* calculate expected arrival time of the next seqnum */
2222 expected = dts + priv->packet_spacing;
2224 delay = get_rtx_delay (priv);
2226 /* and update/install timer for next seqnum */
2228 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2231 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2232 expected, delay, priv->packet_spacing);
2234 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2235 /* if we had a timer, remove it, we don't know when to expect the next
2237 remove_timer (jitterbuffer, timer);
2242 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2245 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2247 /* we need consecutive seqnums with a different
2248 * rtptime to estimate the packet spacing. */
2249 if (priv->ips_rtptime != rtptime) {
2250 /* rtptime changed, check dts diff */
2251 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2252 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2253 GstClockTime old_packet_spacing = priv->packet_spacing;
2255 /* Biased towards bigger packet spacings to prevent
2256 * too many unneeded retransmission requests for next
2257 * packets that just arrive a little later than we would
2259 if (old_packet_spacing > new_packet_spacing)
2260 priv->packet_spacing =
2261 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2262 else if (old_packet_spacing > 0)
2263 priv->packet_spacing =
2264 (3 * new_packet_spacing + old_packet_spacing) / 4;
2266 priv->packet_spacing = new_packet_spacing;
2268 GST_DEBUG_OBJECT (jitterbuffer,
2269 "new packet spacing %" GST_TIME_FORMAT
2270 " old packet spacing %" GST_TIME_FORMAT
2271 " combined to %" GST_TIME_FORMAT,
2272 GST_TIME_ARGS (new_packet_spacing),
2273 GST_TIME_ARGS (old_packet_spacing),
2274 GST_TIME_ARGS (priv->packet_spacing));
2276 priv->ips_rtptime = rtptime;
2277 priv->ips_dts = dts;
2282 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2283 guint16 seqnum, GstClockTime dts, gint gap)
2285 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2286 GstClockTime total_duration, duration, expected_dts;
2289 GST_DEBUG_OBJECT (jitterbuffer,
2290 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2291 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2293 if (dts == GST_CLOCK_TIME_NONE) {
2294 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2298 /* the total duration spanned by the missing packets */
2299 if (dts >= priv->last_in_dts)
2300 total_duration = dts - priv->last_in_dts;
2304 /* interpolate between the current time and the last time based on
2305 * number of packets we are missing, this is the estimated duration
2306 * for the missing packet based on equidistant packet spacing. */
2307 duration = total_duration / (gap + 1);
2309 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2310 GST_TIME_ARGS (duration));
2312 if (total_duration > priv->latency_ns) {
2313 GstClockTime gap_time;
2317 GstClockTime gap_dur = gap * duration;
2318 if (gap_dur > priv->latency_ns)
2319 gap_time = gap_dur - priv->latency_ns;
2322 lost_packets = gap_time / duration;
2324 gap_time = total_duration - priv->latency_ns;
2328 /* too many lost packets, some of the missing packets are already
2329 * too late and we can generate lost packet events for them. */
2330 GST_DEBUG_OBJECT (jitterbuffer,
2331 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2332 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2333 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2334 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2335 GST_TIME_ARGS (gap_time));
2337 /* this timer will fire immediately and the lost event will be pushed from
2338 * the timer thread */
2339 if (lost_packets > 0) {
2340 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2341 priv->last_in_dts + duration, 0, gap_time);
2342 expected += lost_packets;
2343 priv->last_in_dts += gap_time;
2347 expected_dts = priv->last_in_dts + duration;
2349 if (priv->do_retransmission) {
2350 TimerData *timer = find_timer (jitterbuffer, expected);
2352 type = TIMER_TYPE_EXPECTED;
2353 /* if we had a timer for the first missing packet, update it. */
2354 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2355 GstClockTime timeout = timer->timeout;
2357 timer->duration = duration;
2358 if (timeout > (expected_dts + timer->rtx_retry)) {
2359 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2360 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2364 expected_dts += duration;
2367 type = TIMER_TYPE_LOST;
2370 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2371 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2372 expected_dts += duration;
2378 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2382 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2383 GstRtpJitterBufferPrivate *priv;
2385 priv = jitterbuffer->priv;
2387 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2390 if (priv->last_dts != -1)
2391 dtsdiff = dts - priv->last_dts;
2395 if (priv->last_rtptime != -1)
2396 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2400 priv->last_dts = dts;
2401 priv->last_rtptime = rtptime;
2405 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2408 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2410 diff = ABS (dtsdiff - rtpdiffns);
2412 /* jitter is stored in nanoseconds */
2413 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2415 GST_LOG_OBJECT (jitterbuffer,
2416 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2417 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2418 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2419 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2426 GST_DEBUG_OBJECT (jitterbuffer,
2427 "no dts or no clock-rate, can't calculate jitter");
2433 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2435 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2436 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2439 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2440 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2441 gst_rtp_buffer_unmap (&rtp_a);
2443 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2444 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2445 gst_rtp_buffer_unmap (&rtp_b);
2447 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2451 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2452 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout,
2455 GstRtpJitterBufferPrivate *priv;
2456 guint gap_packets_length;
2457 gboolean reset = FALSE;
2459 priv = jitterbuffer->priv;
2461 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2463 guint32 prev_gap_seq = -1;
2464 gboolean all_consecutive = TRUE;
2466 g_queue_insert_sorted (&priv->gap_packets, buffer,
2467 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2469 for (l = priv->gap_packets.head; l; l = l->next) {
2470 GstBuffer *gap_buffer = l->data;
2471 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2474 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2476 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2478 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2479 if (prev_gap_seq == -1)
2480 prev_gap_seq = gap_seq;
2481 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2482 all_consecutive = FALSE;
2484 prev_gap_seq = gap_seq;
2486 gst_rtp_buffer_unmap (&gap_rtp);
2487 if (!all_consecutive)
2491 if (all_consecutive && gap_packets_length > 3) {
2492 GST_DEBUG_OBJECT (jitterbuffer,
2493 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2494 (future ? "new" : "old"), gap,
2495 (future ? max_dropout : -max_misorder));
2497 } else if (!all_consecutive) {
2498 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2499 g_queue_clear (&priv->gap_packets);
2500 GST_DEBUG_OBJECT (jitterbuffer,
2501 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2502 (future ? "new" : "old"), gap,
2503 (future ? max_dropout : -max_misorder));
2506 GST_DEBUG_OBJECT (jitterbuffer,
2507 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2508 (future ? "new" : "old"), gap,
2509 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2513 GST_DEBUG_OBJECT (jitterbuffer,
2514 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2515 gap, -max_misorder);
2516 g_queue_push_tail (&priv->gap_packets, buffer);
2524 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2526 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2527 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2530 GstClockTime base_time =
2531 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2532 GstClockTime clock_time = gst_clock_get_time (clock);
2534 if (clock_time > base_time)
2535 running_time = clock_time - base_time;
2539 gst_object_unref (clock);
2542 return running_time;
2545 static GstFlowReturn
2546 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2549 GstRtpJitterBuffer *jitterbuffer;
2550 GstRtpJitterBufferPrivate *priv;
2552 guint32 expected, rtptime;
2553 GstFlowReturn ret = GST_FLOW_OK;
2554 GstClockTime dts, pts;
2559 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2560 gboolean do_next_seqnum = FALSE;
2561 RTPJitterBufferItem *item;
2562 GstMessage *msg = NULL;
2563 gboolean estimated_dts = FALSE;
2564 guint32 packet_rate, max_dropout, max_misorder;
2566 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2568 priv = jitterbuffer->priv;
2570 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2571 goto invalid_buffer;
2573 pt = gst_rtp_buffer_get_payload_type (&rtp);
2574 seqnum = gst_rtp_buffer_get_seq (&rtp);
2575 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2576 gst_rtp_buffer_unmap (&rtp);
2578 /* make sure we have PTS and DTS set */
2579 pts = GST_BUFFER_PTS (buffer);
2580 dts = GST_BUFFER_DTS (buffer);
2587 /* If we have no DTS here, i.e. no capture time, get one from the
2588 * clock now to have something to calculate with in the future. */
2589 dts = get_current_running_time (jitterbuffer);
2592 /* Remember that we estimated the DTS if we are running already
2593 * and this is not our first packet (or first packet after a reset).
2594 * If it's the first packet, we somehow must generate a timestamp for
2595 * everything, otherwise we can't calculate any times
2597 estimated_dts = (priv->next_in_seqnum != -1);
2599 /* take the DTS of the buffer. This is the time when the packet was
2600 * received and is used to calculate jitter and clock skew. We will adjust
2601 * this DTS with the smoothed value after processing it in the
2602 * jitterbuffer and assign it as the PTS. */
2603 /* bring to running time */
2604 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2607 GST_DEBUG_OBJECT (jitterbuffer,
2608 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2609 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2611 JBUF_LOCK_CHECK (priv, out_flushing);
2613 if (G_UNLIKELY (priv->last_pt != pt)) {
2616 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2620 /* reset clock-rate so that we get a new one */
2621 priv->clock_rate = -1;
2623 /* Try to get the clock-rate from the caps first if we can. If there are no
2624 * caps we must fire the signal to get the clock-rate. */
2625 if ((caps = gst_pad_get_current_caps (pad))) {
2626 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2627 gst_caps_unref (caps);
2631 if (G_UNLIKELY (priv->clock_rate == -1)) {
2632 /* no clock rate given on the caps, try to get one with the signal */
2633 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2634 pt) == GST_FLOW_FLUSHING)
2637 if (G_UNLIKELY (priv->clock_rate == -1))
2640 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
2643 /* don't accept more data on EOS */
2644 if (G_UNLIKELY (priv->eos))
2647 calculate_jitter (jitterbuffer, dts, rtptime);
2649 if (priv->seqnum_base != -1) {
2652 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2655 GST_DEBUG_OBJECT (jitterbuffer,
2656 "packet seqnum #%d before seqnum-base #%d", seqnum,
2658 gst_buffer_unref (buffer);
2661 } else if (gap > 16384) {
2662 /* From now on don't compare against the seqnum base anymore as
2663 * at some point in the future we will wrap around and also that
2664 * much reordering is very unlikely */
2665 priv->seqnum_base = -1;
2669 expected = priv->next_in_seqnum;
2672 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2674 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2675 priv->max_dropout_time);
2677 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2678 priv->max_misorder_time);
2679 GST_TRACE_OBJECT (jitterbuffer,
2680 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2681 max_dropout, max_misorder);
2683 /* now check against our expected seqnum */
2684 if (G_LIKELY (expected != -1)) {
2687 /* now calculate gap */
2688 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2690 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2691 expected, seqnum, gap);
2693 if (G_LIKELY (gap == 0)) {
2694 /* packet is expected */
2695 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2696 do_next_seqnum = TRUE;
2698 gboolean reset = FALSE;
2701 /* we received an old packet */
2702 if (G_UNLIKELY (gap != -1 && gap < -max_misorder)) {
2704 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2705 gap, max_dropout, max_misorder);
2708 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2711 /* new packet, we are missing some packets */
2712 if (G_UNLIKELY (priv->timers->len >= max_dropout)) {
2713 /* If we have timers for more than RTP_MAX_DROPOUT packets
2714 * pending this means that we have a huge gap overall. We can
2715 * reset the jitterbuffer at this point because there's
2716 * just too much data missing to be able to do anything
2717 * sensible with the past data. Just try again from the
2719 GST_WARNING_OBJECT (jitterbuffer,
2720 "%d pending timers > %d - resetting", priv->timers->len,
2723 gst_buffer_unref (buffer);
2725 } else if (G_UNLIKELY (gap >= max_dropout)) {
2727 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2728 gap, max_dropout, max_misorder);
2731 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2732 /* fill in the gap with EXPECTED timers */
2733 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2735 do_next_seqnum = TRUE;
2738 if (G_UNLIKELY (reset)) {
2739 GList *events = NULL, *l;
2742 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2743 rtp_jitter_buffer_flush (priv->jbuf,
2744 (GFunc) free_item_and_retain_events, &events);
2745 rtp_jitter_buffer_reset_skew (priv->jbuf);
2746 remove_all_timers (jitterbuffer);
2747 priv->discont = TRUE;
2748 priv->last_popped_seqnum = -1;
2750 if (priv->gap_packets.head) {
2751 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2752 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2754 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2755 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2756 gst_rtp_buffer_unmap (&gap_rtp);
2758 priv->next_seqnum = seqnum;
2761 priv->last_in_dts = -1;
2762 priv->next_in_seqnum = -1;
2764 /* Insert all sticky events again in order, otherwise we would
2765 * potentially loose STREAM_START, CAPS or SEGMENT events
2767 events = g_list_reverse (events);
2768 for (l = events; l; l = l->next) {
2769 RTPJitterBufferItem *item;
2771 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2772 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
2774 g_list_free (events);
2776 JBUF_SIGNAL_EVENT (priv);
2778 /* reset spacing estimation when gap */
2779 priv->ips_rtptime = -1;
2780 priv->ips_dts = GST_CLOCK_TIME_NONE;
2782 buffers = g_list_copy (priv->gap_packets.head);
2783 g_queue_clear (&priv->gap_packets);
2785 priv->ips_rtptime = -1;
2786 priv->ips_dts = GST_CLOCK_TIME_NONE;
2787 JBUF_UNLOCK (jitterbuffer->priv);
2789 for (l = buffers; l; l = l->next) {
2790 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2792 if (ret != GST_FLOW_OK) {
2797 for (; l; l = l->next)
2798 gst_buffer_unref (l->data);
2799 g_list_free (buffers);
2803 /* reset spacing estimation when gap */
2804 priv->ips_rtptime = -1;
2805 priv->ips_dts = GST_CLOCK_TIME_NONE;
2808 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2810 /* we don't know what the next_in_seqnum should be, wait for the last
2811 * possible moment to push this buffer, maybe we get an earlier seqnum
2813 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2814 do_next_seqnum = TRUE;
2815 /* take rtptime and dts to calculate packet spacing */
2816 priv->ips_rtptime = rtptime;
2817 priv->ips_dts = dts;
2820 /* We had no huge gap, let's drop all the gap packets */
2821 if (buffer != NULL) {
2822 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2823 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2824 g_queue_clear (&priv->gap_packets);
2826 GST_DEBUG_OBJECT (jitterbuffer,
2827 "Had big gap, waiting for more consecutive packets");
2828 JBUF_UNLOCK (jitterbuffer->priv);
2832 if (do_next_seqnum) {
2833 priv->last_in_dts = dts;
2834 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2837 /* let's check if this buffer is too late, we can only accept packets with
2838 * bigger seqnum than the one we last pushed. */
2839 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2842 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2844 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2845 if (G_UNLIKELY (gap <= 0))
2849 if (already_lost (jitterbuffer, seqnum))
2852 /* let's drop oldest packet if the queue is already full and drop-on-latency
2853 * is set. We can only do this when there actually is a latency. When no
2854 * latency is set, we just pump it in the queue and let the other end push it
2855 * out as fast as possible. */
2856 if (priv->latency_ms && priv->drop_on_latency) {
2858 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2860 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2861 RTPJitterBufferItem *old_item;
2863 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2865 if (IS_DROPABLE (old_item)) {
2866 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2867 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2869 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2870 free_item (old_item);
2872 /* we might have removed some head buffers, signal the pushing thread to
2873 * see if it can push now */
2874 JBUF_SIGNAL_EVENT (priv);
2878 /* If we estimated the DTS, don't consider it in the clock skew calculations
2879 * later. The code above always sets dts to pts or the other way around if
2880 * any of those is valid in the buffer, so we know that if we estimated the
2881 * dts that both are unknown */
2884 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2885 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2887 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2889 /* now insert the packet into the queue in sorted order. This function returns
2890 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2891 * have a duplicate. */
2892 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2894 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)))))
2898 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2900 /* we had an unhandled SR, handle it now */
2902 do_handle_sync (jitterbuffer);
2904 if (G_UNLIKELY (head)) {
2905 /* signal addition of new buffer when the _loop is waiting. */
2906 if (G_LIKELY (priv->active))
2907 JBUF_SIGNAL_EVENT (priv);
2909 /* let's unschedule and unblock any waiting buffers. We only want to do this
2910 * when the head buffer changed */
2911 if (G_UNLIKELY (priv->clock_id)) {
2912 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2913 unschedule_current_timer (jitterbuffer);
2917 GST_DEBUG_OBJECT (jitterbuffer,
2918 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2919 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2921 msg = check_buffering_percent (jitterbuffer, percent);
2927 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2934 /* this is not fatal but should be filtered earlier */
2935 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2936 ("Received invalid RTP payload, dropping"));
2937 gst_buffer_unref (buffer);
2942 GST_WARNING_OBJECT (jitterbuffer,
2943 "No clock-rate in caps!, dropping buffer");
2944 gst_buffer_unref (buffer);
2949 ret = priv->srcresult;
2950 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2951 gst_buffer_unref (buffer);
2957 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2958 gst_buffer_unref (buffer);
2963 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2964 " popped, dropping", seqnum, priv->last_popped_seqnum);
2966 gst_buffer_unref (buffer);
2971 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
2972 "considered lost", seqnum);
2974 gst_buffer_unref (buffer);
2979 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2981 priv->num_duplicates++;
2988 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2990 guint64 ext_time, elapsed;
2992 GstRtpJitterBufferPrivate *priv;
2994 priv = jitterbuffer->priv;
2995 rtp_time = item->rtptime;
2997 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2998 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3000 ext_time = priv->ext_timestamp;
3001 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3002 if (ext_time < priv->ext_timestamp) {
3003 ext_time = priv->ext_timestamp;
3005 priv->ext_timestamp = ext_time;
3008 if (ext_time > priv->clock_base)
3009 elapsed = ext_time - priv->clock_base;
3013 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3018 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3019 RTPJitterBufferItem * item)
3021 guint64 total, elapsed, left, estimated;
3022 GstClockTime out_time;
3023 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3025 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3026 || priv->clock_base == -1 || priv->clock_rate <= 0)
3029 /* compute the elapsed time */
3030 elapsed = compute_elapsed (jitterbuffer, item);
3032 /* do nothing if elapsed time doesn't increment */
3033 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3036 priv->last_elapsed = elapsed;
3038 /* this is the total time we need to play */
3039 total = priv->npt_stop - priv->npt_start;
3040 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3041 GST_TIME_ARGS (total));
3043 /* this is how much time there is left */
3044 if (total > elapsed)
3045 left = total - elapsed;
3049 /* if we have less time left that the size of the buffer, we will not
3050 * be able to keep it filled, disabled buffering then */
3051 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3052 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3053 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3054 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3057 /* this is the current time as running-time */
3058 out_time = item->dts;
3061 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3063 /* if there is almost nothing left,
3064 * we may never advance enough to end up in the above case */
3065 if (total < GST_SECOND)
3066 estimated = GST_SECOND;
3070 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3071 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3073 if (estimated != -1 && priv->estimated_eos != estimated) {
3074 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3075 priv->estimated_eos = estimated;
3079 /* take a buffer from the queue and push it */
3080 static GstFlowReturn
3081 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3083 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3084 GstFlowReturn result = GST_FLOW_OK;
3085 RTPJitterBufferItem *item;
3086 GstBuffer *outbuf = NULL;
3087 GstEvent *outevent = NULL;
3088 GstQuery *outquery = NULL;
3089 GstClockTime dts, pts;
3091 gboolean do_push = TRUE;
3095 /* when we get here we are ready to pop and push the buffer */
3096 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3100 case ITEM_TYPE_BUFFER:
3102 /* we need to make writable to change the flags and timestamps */
3103 outbuf = gst_buffer_make_writable (item->data);
3105 if (G_UNLIKELY (priv->discont)) {
3106 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3107 * into the jitterbuffer so we can modify now. */
3108 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3109 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3110 priv->discont = FALSE;
3112 if (G_UNLIKELY (priv->ts_discont)) {
3113 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3114 priv->ts_discont = FALSE;
3118 gst_segment_position_from_running_time (&priv->segment,
3119 GST_FORMAT_TIME, item->dts);
3121 gst_segment_position_from_running_time (&priv->segment,
3122 GST_FORMAT_TIME, item->pts);
3124 /* apply timestamp with offset to buffer now */
3125 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3126 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3128 /* update the elapsed time when we need to check against the npt stop time. */
3129 update_estimated_eos (jitterbuffer, item);
3131 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3133 case ITEM_TYPE_LOST:
3134 priv->discont = TRUE;
3138 case ITEM_TYPE_EVENT:
3139 outevent = item->data;
3141 case ITEM_TYPE_QUERY:
3142 outquery = item->data;
3146 /* now we are ready to push the buffer. Save the seqnum and release the lock
3147 * so the other end can push stuff in the queue again. */
3149 priv->last_popped_seqnum = seqnum;
3150 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3152 msg = check_buffering_percent (jitterbuffer, percent);
3159 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3162 case ITEM_TYPE_BUFFER:
3164 GST_DEBUG_OBJECT (jitterbuffer,
3165 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3166 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3167 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3168 result = gst_pad_push (priv->srcpad, outbuf);
3170 JBUF_LOCK_CHECK (priv, out_flushing);
3172 case ITEM_TYPE_LOST:
3173 case ITEM_TYPE_EVENT:
3174 /* We got not enough consecutive packets with a huge gap, we can
3175 * as well just drop them here now on EOS */
3176 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3177 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3178 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3179 g_queue_clear (&priv->gap_packets);
3182 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3183 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3186 gst_pad_push_event (priv->srcpad, outevent);
3188 gst_event_unref (outevent);
3190 result = GST_FLOW_OK;
3192 JBUF_LOCK_CHECK (priv, out_flushing);
3194 case ITEM_TYPE_QUERY:
3198 res = gst_pad_peer_query (priv->srcpad, outquery);
3200 JBUF_LOCK_CHECK (priv, out_flushing);
3201 result = GST_FLOW_OK;
3202 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3203 JBUF_SIGNAL_QUERY (priv, res);
3212 return priv->srcresult;
3216 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3218 /* Peek a buffer and compare the seqnum to the expected seqnum.
3219 * If all is fine, the buffer is pushed.
3220 * If something is wrong, we wait for some event
3222 static GstFlowReturn
3223 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3225 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3226 GstFlowReturn result;
3227 RTPJitterBufferItem *item;
3229 guint32 next_seqnum;
3231 /* only push buffers when PLAYING and active and not buffering */
3232 if (priv->blocked || !priv->active ||
3233 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3234 return GST_FLOW_WAIT;
3237 /* peek a buffer, we're just looking at the sequence number.
3238 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3239 * wait for a timeout or something to change.
3240 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3241 item = rtp_jitter_buffer_peek (priv->jbuf);
3246 /* get the seqnum and the next expected seqnum */
3247 seqnum = item->seqnum;
3249 return pop_and_push_next (jitterbuffer, seqnum);
3252 next_seqnum = priv->next_seqnum;
3254 /* get the gap between this and the previous packet. If we don't know the
3255 * previous packet seqnum assume no gap. */
3256 if (G_UNLIKELY (next_seqnum == -1)) {
3257 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3258 /* we don't know what the next_seqnum should be, the chain function should
3259 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3260 * fires, so wait for that */
3261 result = GST_FLOW_WAIT;
3263 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3265 if (G_LIKELY (gap == 0)) {
3266 /* no missing packet, pop and push */
3267 result = pop_and_push_next (jitterbuffer, seqnum);
3268 } else if (G_UNLIKELY (gap < 0)) {
3269 /* if we have a packet that we already pushed or considered dropped, pop it
3270 * off and get the next packet */
3271 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3272 seqnum, next_seqnum);
3273 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3275 result = GST_FLOW_OK;
3277 /* the chain function has scheduled timers to request retransmission or
3278 * when to consider the packet lost, wait for that */
3279 GST_DEBUG_OBJECT (jitterbuffer,
3280 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3281 next_seqnum, seqnum, gap);
3282 result = GST_FLOW_WAIT;
3290 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3292 return GST_FLOW_EOS;
3294 return GST_FLOW_WAIT;
3300 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3302 GstClockTime rtx_retry_timeout;
3303 GstClockTime rtx_min_retry_timeout;
3305 if (priv->rtx_retry_timeout == -1) {
3306 if (priv->avg_rtx_rtt == 0)
3307 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3309 /* we want to ask for a retransmission after we waited for a
3310 * complete RTT and the additional jitter */
3311 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3313 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3315 /* make sure we don't retry too often. On very low latency networks,
3316 * the RTT and jitter can be very low. */
3317 if (priv->rtx_min_retry_timeout == -1) {
3318 rtx_min_retry_timeout = priv->packet_spacing;
3320 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3322 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3324 return rtx_retry_timeout;
3328 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3329 GstClockTime rtx_retry_timeout)
3331 GstClockTime rtx_retry_period;
3333 if (priv->rtx_retry_period == -1) {
3334 /* we retry up to the configured jitterbuffer size but leaving some
3335 * room for the retransmission to arrive in time */
3336 if (rtx_retry_timeout > priv->latency_ns) {
3337 rtx_retry_period = 0;
3339 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3342 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3344 return rtx_retry_period;
3347 /* the timeout for when we expected a packet expired */
3349 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3352 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3354 guint delay, delay_ms, avg_rtx_rtt_ms;
3355 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3356 GstClockTime rtx_retry_period;
3357 GstClockTime rtx_retry_timeout;
3360 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3361 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3363 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3364 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3366 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3367 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3368 GST_TIME_ARGS (rtx_retry_period));
3370 delay = timer->rtx_delay + timer->rtx_retry;
3372 delay_ms = GST_TIME_AS_MSECONDS (delay);
3373 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3374 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3375 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3377 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3378 gst_structure_new ("GstRTPRetransmissionRequest",
3379 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3380 "running-time", G_TYPE_UINT64, timer->rtx_base,
3381 "delay", G_TYPE_UINT, delay_ms,
3382 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3383 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3384 "period", G_TYPE_UINT, rtx_retry_period_ms,
3385 "deadline", G_TYPE_UINT, priv->latency_ms,
3386 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3387 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3389 priv->num_rtx_requests++;
3390 timer->num_rtx_retry++;
3392 GST_OBJECT_LOCK (jitterbuffer);
3393 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3394 timer->rtx_last = gst_clock_get_time (clock);
3395 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3397 timer->rtx_last = now;
3399 GST_OBJECT_UNLOCK (jitterbuffer);
3401 /* calculate the timeout for the next retransmission attempt */
3402 timer->rtx_retry += rtx_retry_timeout;
3403 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3404 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3405 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3406 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3407 if ((priv->rtx_max_retries != -1
3408 && timer->num_rtx_retry >= priv->rtx_max_retries)
3409 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3410 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3411 /* too many retransmission request, we now convert the timer
3412 * to a lost timer, leave the num_rtx_retry as it is for stats */
3413 timer->type = TIMER_TYPE_LOST;
3414 timer->rtx_delay = 0;
3415 timer->rtx_retry = 0;
3417 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3418 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3421 gst_pad_push_event (priv->sinkpad, event);
3427 /* a packet is lost */
3429 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3432 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3433 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3435 GstEvent *event = NULL;
3436 RTPJitterBufferItem *item;
3438 seqnum = timer->seqnum;
3439 lost_packets = MAX (timer->num, 1);
3440 num_rtx_retry = timer->num_rtx_retry;
3442 /* we had a gap and thus we lost some packets. Create an event for this. */
3443 if (lost_packets > 1)
3444 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3445 seqnum + lost_packets - 1);
3447 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3449 priv->num_late += lost_packets;
3450 priv->num_rtx_failed += num_rtx_retry;
3452 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3454 /* we now only accept seqnum bigger than this */
3455 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3456 priv->next_in_seqnum = next_in_seqnum;
3458 /* Avoid creating events if we don't need it. Note that we still need to create
3459 * the lost *ITEM* since it will be used to notify the outgoing thread of
3460 * lost items (so that we can set discont flags and such) */
3461 if (priv->do_lost) {
3462 GstClockTime duration, timestamp;
3463 /* create paket lost event */
3464 timestamp = apply_offset (jitterbuffer, timer->timeout);
3465 duration = timer->duration;
3466 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3467 duration = priv->packet_spacing;
3468 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3469 gst_structure_new ("GstRTPPacketLost",
3470 "seqnum", G_TYPE_UINT, (guint) seqnum,
3471 "timestamp", G_TYPE_UINT64, timestamp,
3472 "duration", G_TYPE_UINT64, duration,
3473 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3475 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3476 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3478 /* remove timer now */
3479 remove_timer (jitterbuffer, timer);
3481 JBUF_SIGNAL_EVENT (priv);
3487 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3490 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3492 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3493 remove_timer (jitterbuffer, timer);
3495 /* there was no EOS in the buffer, put one in there now */
3496 queue_event (jitterbuffer, gst_event_new_eos ());
3498 JBUF_SIGNAL_EVENT (priv);
3504 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3507 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3509 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3511 /* timer seqnum might have been obsoleted by caps seqnum-base,
3512 * only mess with current ongoing seqnum if still unknown */
3513 if (priv->next_seqnum == -1)
3514 priv->next_seqnum = timer->seqnum;
3515 remove_timer (jitterbuffer, timer);
3516 JBUF_SIGNAL_EVENT (priv);
3522 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3525 gboolean removed = FALSE;
3527 switch (timer->type) {
3528 case TIMER_TYPE_EXPECTED:
3529 removed = do_expected_timeout (jitterbuffer, timer, now);
3531 case TIMER_TYPE_LOST:
3532 removed = do_lost_timeout (jitterbuffer, timer, now);
3534 case TIMER_TYPE_DEADLINE:
3535 removed = do_deadline_timeout (jitterbuffer, timer, now);
3537 case TIMER_TYPE_EOS:
3538 removed = do_eos_timeout (jitterbuffer, timer, now);
3544 /* called when we need to wait for the next timeout.
3546 * We loop over the array of recorded timeouts and wait for the earliest one.
3547 * When it timed out, do the logic associated with the timer.
3549 * If there are no timers, we wait on a gcond until something new happens.
3552 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3554 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3555 GstClockTime now = 0;
3558 while (priv->timer_running) {
3559 TimerData *timer = NULL;
3560 GstClockTime timer_timeout = -1;
3563 /* If we have a clock, update "now" now with the very
3564 * latest running time we have. If timers are unscheduled below we
3565 * otherwise wouldn't update now (it's only updated when timers
3566 * expire), and also for the very first loop iteration now would
3567 * otherwise always be 0
3569 GST_OBJECT_LOCK (jitterbuffer);
3570 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3572 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3573 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3575 GST_OBJECT_UNLOCK (jitterbuffer);
3577 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3578 GST_TIME_ARGS (now));
3580 len = priv->timers->len;
3581 for (i = 0; i < len;) {
3582 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3583 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3584 gboolean save_best = FALSE;
3586 GST_DEBUG_OBJECT (jitterbuffer,
3587 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
3588 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
3589 GST_STIME_ARGS ((gint64) (test_timeout - now)));
3591 /* Weed out anything too late */
3592 if (test->type == TIMER_TYPE_LOST &&
3593 (test_timeout == -1 || test_timeout <= now)) {
3594 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
3595 do_lost_timeout (jitterbuffer, test, now);
3596 if (!priv->timer_running)
3598 /* We don't move the iterator forward since we just removed the current entry,
3599 * but we update the termination condition */
3600 len = priv->timers->len;
3602 /* find the smallest timeout */
3603 if (timer == NULL) {
3605 } else if (timer_timeout == -1) {
3606 /* we already have an immediate timeout, the new timer must be an
3607 * immediate timer with smaller seqnum to become the best */
3608 if (test_timeout == -1
3609 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3610 timer->seqnum) > 0))
3612 } else if (test_timeout == -1) {
3613 /* first immediate timer */
3615 } else if (test_timeout < timer_timeout) {
3618 } else if (test_timeout == timer_timeout
3619 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3620 timer->seqnum) > 0)) {
3621 /* same timer, smaller seqnum */
3626 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3628 timer_timeout = test_timeout;
3633 if (timer && !priv->blocked) {
3635 GstClockTime sync_time;
3638 GstClockTimeDiff clock_jitter;
3640 if (timer_timeout == -1 || timer_timeout <= now) {
3641 /* We have normally removed all lost timers in the loop above */
3642 g_assert (timer->type != TIMER_TYPE_LOST);
3644 do_timeout (jitterbuffer, timer, now);
3645 /* check here, do_timeout could have released the lock */
3646 if (!priv->timer_running)
3651 GST_OBJECT_LOCK (jitterbuffer);
3652 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3654 GST_OBJECT_UNLOCK (jitterbuffer);
3655 /* let's just push if there is no clock */
3656 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3657 now = timer_timeout;
3661 /* prepare for sync against clock */
3662 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3663 /* add latency of peer to get input time */
3664 sync_time += priv->peer_latency;
3666 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3667 " with sync time %" GST_TIME_FORMAT,
3668 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3670 /* create an entry for the clock */
3671 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3672 priv->timer_timeout = timer_timeout;
3673 priv->timer_seqnum = timer->seqnum;
3674 GST_OBJECT_UNLOCK (jitterbuffer);
3676 /* release the lock so that the other end can push stuff or unlock */
3679 ret = gst_clock_id_wait (id, &clock_jitter);
3682 if (!priv->timer_running) {
3683 gst_clock_id_unref (id);
3684 priv->clock_id = NULL;
3688 if (ret != GST_CLOCK_UNSCHEDULED) {
3689 now = timer_timeout + MAX (clock_jitter, 0);
3690 GST_DEBUG_OBJECT (jitterbuffer,
3691 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3692 GST_STIME_ARGS (clock_jitter));
3694 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3696 /* and free the entry */
3697 gst_clock_id_unref (id);
3698 priv->clock_id = NULL;
3700 /* no timers, wait for activity */
3701 JBUF_WAIT_TIMER (priv);
3706 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3711 * This funcion implements the main pushing loop on the source pad.
3713 * It first tries to push as many buffers as possible. If there is a seqnum
3714 * mismatch, we wait for the next timeouts.
3717 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3719 GstRtpJitterBufferPrivate *priv;
3720 GstFlowReturn result = GST_FLOW_OK;
3722 priv = jitterbuffer->priv;
3724 JBUF_LOCK_CHECK (priv, flushing);
3726 result = handle_next_buffer (jitterbuffer);
3727 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3728 /* now wait for the next event */
3729 JBUF_WAIT_EVENT (priv, flushing);
3730 result = GST_FLOW_OK;
3732 } while (result == GST_FLOW_OK);
3733 /* store result for upstream */
3734 priv->srcresult = result;
3735 /* if we get here we need to pause */
3741 result = priv->srcresult;
3748 JBUF_SIGNAL_QUERY (priv, FALSE);
3751 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3752 gst_flow_get_name (result));
3753 gst_pad_pause_task (priv->srcpad);
3754 if (result == GST_FLOW_EOS) {
3755 event = gst_event_new_eos ();
3756 gst_pad_push_event (priv->srcpad, event);
3762 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3763 * some sanity checks and then emit the handle-sync signal with the parameters.
3764 * This function must be called with the LOCK */
3766 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3768 GstRtpJitterBufferPrivate *priv;
3769 guint64 base_rtptime, base_time;
3771 guint64 last_rtptime;
3773 guint64 ext_rtptime, diff;
3774 gboolean valid = TRUE, keep = FALSE;
3776 priv = jitterbuffer->priv;
3778 /* get the last values from the jitterbuffer */
3779 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3780 &clock_rate, &last_rtptime);
3782 clock_base = priv->clock_base;
3783 ext_rtptime = priv->ext_rtptime;
3785 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3786 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3787 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3788 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3790 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3791 /* we keep this SR packet for later. When we get a valid RTP packet the
3792 * above values will be set and we can try to use the SR packet */
3793 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3796 /* we can't accept anything that happened before we did the last resync */
3797 if (base_rtptime > ext_rtptime) {
3798 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3801 /* the SR RTP timestamp must be something close to what we last observed
3802 * in the jitterbuffer */
3803 if (ext_rtptime > last_rtptime) {
3804 /* check how far ahead it is to our RTP timestamps */
3805 diff = ext_rtptime - last_rtptime;
3806 /* if bigger than 1 second, we drop it */
3807 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
3809 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
3810 clock_rate, 1000)) {
3811 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3812 /* should drop this, but some RTSP servers end up with bogus
3813 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3814 * so still trigger rptbin sync but invalidate RTCP data
3815 * (sync might use other methods) */
3818 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3819 G_GUINT64_FORMAT, last_rtptime, diff);
3825 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3829 s = gst_structure_new ("application/x-rtp-sync",
3830 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3831 "base-time", G_TYPE_UINT64, base_time,
3832 "clock-rate", G_TYPE_UINT, clock_rate,
3833 "clock-base", G_TYPE_UINT64, clock_base,
3834 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3835 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3837 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3838 gst_buffer_replace (&priv->last_sr, NULL);
3840 g_signal_emit (jitterbuffer,
3841 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3843 gst_structure_free (s);
3845 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3846 gst_buffer_replace (&priv->last_sr, NULL);
3850 static GstFlowReturn
3851 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3854 GstRtpJitterBuffer *jitterbuffer;
3855 GstRtpJitterBufferPrivate *priv;
3856 GstFlowReturn ret = GST_FLOW_OK;
3858 GstRTCPPacket packet;
3859 guint64 ext_rtptime;
3861 GstRTCPBuffer rtcp = { NULL, };
3863 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3865 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3866 goto invalid_buffer;
3868 priv = jitterbuffer->priv;
3870 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3872 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3875 /* first packet must be SR or RR or else the validate would have failed */
3876 switch (gst_rtcp_packet_get_type (&packet)) {
3877 case GST_RTCP_TYPE_SR:
3878 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3884 gst_rtcp_buffer_unmap (&rtcp);
3886 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3889 /* convert the RTP timestamp to our extended timestamp, using the same offset
3890 * we used in the jitterbuffer */
3891 ext_rtptime = priv->jbuf->ext_rtptime;
3892 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3894 priv->ext_rtptime = ext_rtptime;
3895 gst_buffer_replace (&priv->last_sr, buffer);
3897 do_handle_sync (jitterbuffer);
3901 gst_buffer_unref (buffer);
3907 /* this is not fatal but should be filtered earlier */
3908 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3909 ("Received invalid RTCP payload, dropping"));
3915 /* this is not fatal but should be filtered earlier */
3916 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3917 ("Received empty RTCP payload, dropping"));
3918 gst_rtcp_buffer_unmap (&rtcp);
3924 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3925 gst_rtcp_buffer_unmap (&rtcp);
3932 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3935 gboolean res = FALSE;
3936 GstRtpJitterBuffer *jitterbuffer;
3937 GstRtpJitterBufferPrivate *priv;
3939 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3940 priv = jitterbuffer->priv;
3942 switch (GST_QUERY_TYPE (query)) {
3943 case GST_QUERY_CAPS:
3945 GstCaps *filter, *caps;
3947 gst_query_parse_caps (query, &filter);
3948 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3949 gst_query_set_caps_result (query, caps);
3950 gst_caps_unref (caps);
3955 if (GST_QUERY_IS_SERIALIZED (query)) {
3956 RTPJitterBufferItem *item;
3959 JBUF_LOCK_CHECK (priv, out_flushing);
3960 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3961 RTP_JITTER_BUFFER_MODE_BUFFER) {
3962 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3963 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3964 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3966 JBUF_SIGNAL_EVENT (priv);
3967 JBUF_WAIT_QUERY (priv, out_flushing);
3968 res = priv->last_query;
3970 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3975 res = gst_pad_query_default (pad, parent, query);
3983 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3991 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3994 GstRtpJitterBuffer *jitterbuffer;
3995 GstRtpJitterBufferPrivate *priv;
3996 gboolean res = FALSE;
3998 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3999 priv = jitterbuffer->priv;
4001 switch (GST_QUERY_TYPE (query)) {
4002 case GST_QUERY_LATENCY:
4004 /* We need to send the query upstream and add the returned latency to our
4006 GstClockTime min_latency, max_latency;
4008 GstClockTime our_latency;
4010 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4011 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4013 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4014 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4015 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4017 /* store this so that we can safely sync on the peer buffers. */
4019 priv->peer_latency = min_latency;
4020 our_latency = priv->latency_ns;
4023 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4024 GST_TIME_ARGS (our_latency));
4026 /* we add some latency but can buffer an infinite amount of time */
4027 min_latency += our_latency;
4030 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4031 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4032 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4034 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4038 case GST_QUERY_POSITION:
4040 GstClockTime start, last_out;
4043 gst_query_parse_position (query, &fmt, NULL);
4044 if (fmt != GST_FORMAT_TIME) {
4045 res = gst_pad_query_default (pad, parent, query);
4050 start = priv->npt_start;
4051 last_out = priv->last_out_time;
4054 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4055 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4056 GST_TIME_ARGS (last_out));
4058 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4059 /* bring 0-based outgoing time to stream time */
4060 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4063 res = gst_pad_query_default (pad, parent, query);
4067 case GST_QUERY_CAPS:
4069 GstCaps *filter, *caps;
4071 gst_query_parse_caps (query, &filter);
4072 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4073 gst_query_set_caps_result (query, caps);
4074 gst_caps_unref (caps);
4079 res = gst_pad_query_default (pad, parent, query);
4087 gst_rtp_jitter_buffer_set_property (GObject * object,
4088 guint prop_id, const GValue * value, GParamSpec * pspec)
4090 GstRtpJitterBuffer *jitterbuffer;
4091 GstRtpJitterBufferPrivate *priv;
4093 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4094 priv = jitterbuffer->priv;
4099 guint new_latency, old_latency;
4101 new_latency = g_value_get_uint (value);
4104 old_latency = priv->latency_ms;
4105 priv->latency_ms = new_latency;
4106 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4107 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4110 /* post message if latency changed, this will inform the parent pipeline
4111 * that a latency reconfiguration is possible/needed. */
4112 if (new_latency != old_latency) {
4113 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4114 GST_TIME_ARGS (new_latency * GST_MSECOND));
4116 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4117 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4121 case PROP_DROP_ON_LATENCY:
4123 priv->drop_on_latency = g_value_get_boolean (value);
4126 case PROP_TS_OFFSET:
4128 priv->ts_offset = g_value_get_int64 (value);
4129 priv->ts_discont = TRUE;
4134 priv->do_lost = g_value_get_boolean (value);
4139 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4142 case PROP_DO_RETRANSMISSION:
4144 priv->do_retransmission = g_value_get_boolean (value);
4147 case PROP_RTX_NEXT_SEQNUM:
4149 priv->rtx_next_seqnum = g_value_get_boolean (value);
4152 case PROP_RTX_DELAY:
4154 priv->rtx_delay = g_value_get_int (value);
4157 case PROP_RTX_MIN_DELAY:
4159 priv->rtx_min_delay = g_value_get_uint (value);
4162 case PROP_RTX_DELAY_REORDER:
4164 priv->rtx_delay_reorder = g_value_get_int (value);
4167 case PROP_RTX_RETRY_TIMEOUT:
4169 priv->rtx_retry_timeout = g_value_get_int (value);
4172 case PROP_RTX_MIN_RETRY_TIMEOUT:
4174 priv->rtx_min_retry_timeout = g_value_get_int (value);
4177 case PROP_RTX_RETRY_PERIOD:
4179 priv->rtx_retry_period = g_value_get_int (value);
4182 case PROP_RTX_MAX_RETRIES:
4184 priv->rtx_max_retries = g_value_get_int (value);
4187 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4189 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4192 case PROP_MAX_DROPOUT_TIME:
4194 priv->max_dropout_time = g_value_get_uint (value);
4197 case PROP_MAX_MISORDER_TIME:
4199 priv->max_misorder_time = g_value_get_uint (value);
4202 case PROP_RFC7273_SYNC:
4204 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4205 g_value_get_boolean (value));
4209 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4215 gst_rtp_jitter_buffer_get_property (GObject * object,
4216 guint prop_id, GValue * value, GParamSpec * pspec)
4218 GstRtpJitterBuffer *jitterbuffer;
4219 GstRtpJitterBufferPrivate *priv;
4221 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4222 priv = jitterbuffer->priv;
4227 g_value_set_uint (value, priv->latency_ms);
4230 case PROP_DROP_ON_LATENCY:
4232 g_value_set_boolean (value, priv->drop_on_latency);
4235 case PROP_TS_OFFSET:
4237 g_value_set_int64 (value, priv->ts_offset);
4242 g_value_set_boolean (value, priv->do_lost);
4247 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4255 if (priv->srcresult != GST_FLOW_OK)
4258 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4260 g_value_set_int (value, percent);
4264 case PROP_DO_RETRANSMISSION:
4266 g_value_set_boolean (value, priv->do_retransmission);
4269 case PROP_RTX_NEXT_SEQNUM:
4271 g_value_set_boolean (value, priv->rtx_next_seqnum);
4274 case PROP_RTX_DELAY:
4276 g_value_set_int (value, priv->rtx_delay);
4279 case PROP_RTX_MIN_DELAY:
4281 g_value_set_uint (value, priv->rtx_min_delay);
4284 case PROP_RTX_DELAY_REORDER:
4286 g_value_set_int (value, priv->rtx_delay_reorder);
4289 case PROP_RTX_RETRY_TIMEOUT:
4291 g_value_set_int (value, priv->rtx_retry_timeout);
4294 case PROP_RTX_MIN_RETRY_TIMEOUT:
4296 g_value_set_int (value, priv->rtx_min_retry_timeout);
4299 case PROP_RTX_RETRY_PERIOD:
4301 g_value_set_int (value, priv->rtx_retry_period);
4304 case PROP_RTX_MAX_RETRIES:
4306 g_value_set_int (value, priv->rtx_max_retries);
4310 g_value_take_boxed (value,
4311 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4313 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4315 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4318 case PROP_MAX_DROPOUT_TIME:
4320 g_value_set_uint (value, priv->max_dropout_time);
4323 case PROP_MAX_MISORDER_TIME:
4325 g_value_set_uint (value, priv->max_misorder_time);
4328 case PROP_RFC7273_SYNC:
4330 g_value_set_boolean (value,
4331 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4335 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4340 static GstStructure *
4341 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4345 JBUF_LOCK (jbuf->priv);
4346 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4347 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4348 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4349 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4350 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4351 JBUF_UNLOCK (jbuf->priv);